It is well known that large rooms are a challenging environment for acoustic communication. For example, in a typical classroom the distance between the teacher and the students is considerably larger than the typical distance of about 1 to 1.5 meter during a normal one-to-one communication. Therefore, the voice of the teacher is relatively weak at the position of the student. External noise sources as well as the voices or other sounds coming from fellow students add to the acoustics, resulting in a low or even negative signal-to-noise ratio. In addition, the walls of the room add acoustic reverberation to the voice of the teacher, which further reduces its intelligibility.
While the above problems are especially severe for hearing impaired students in a classroom, they are also present outside the classroom, e.g. during a business meeting in a large room, in a room with poor room acoustics or simply because some meeting participants talk or make sounds during a presentation.
In the prior art a solution is implemented based on radio transmission of the teacher's voice to the hearing aids of the student. This system is referred to as an “FM system”. FIG. 1 provides an illustration. By means of FM wireless transmission, audio data is transferred to a receiver, which then plays the signal to a headphone or hearing aid.
In recent years FM systems have been improved by implementing an estimation of the signal-to-noise ratio in separate frequency regions at the position of the teacher and transmitting this information to the FM receiver in order to improve speech intelligibility by adding gain to the voice of the teacher in frequency regions with more noise energy.
An important limitation of FM systems is that they don't offer a way to change the signal latency. This is especially critical in applications where latency needs to be increased to be in sync with, for example, a video stream. Another important limitation is that FM systems are analog, offering no way at the end point to perform signal error correction, and they are susceptible to interference. Those systems are also unidirectional, making it impossible to transmit back audio from the receiver using the same frequency band.
An important challenge in any digital audio solution is to keep the latency introduced by the digital signal processing as low as possible. The latency of a system is defined as the time difference between the time at which some data is received in the system and the time of which the same data is outputted. The challenge imposed by WiFi technology when it comes to low latency audio, is well known. To our knowledge, no audio-over-network solution available on the market offers an audio latency over WiFi of less than 100 ms. Impairments like jitter, radio mode change or other are very detrimental when it comes to a low latency stream of data.
VoIP solutions can be applied over WiFi to obtain a system capable of streaming real time audio over a local area network. Such systems, however, are generally designed to communicate audio over the Internet and have less restrictions on latency requirements, as they normally interconnect people that do not have direct visual contact because they communicate over larger distance and are not in the same room. Because of the large latency, these systems are not generally suitable for use to transmit audio on a latency-constrained environment such as for communication in the same room.
Similar observations can be made with respect to video data. The importance of low latency can be illustrated for the case of a deaf person who wants to follow a conference at which he is physically present and where an additional video stream is broadcasted to an assistive device (like a smartphone, smartglass, etc.). The same latency-constrained environment occurs in a concert, where the audience receives the audio signal directly from the public address system, but multiple video streams are available for those who want to see on their personal communication devices details of the concert they don't want to miss (for example, a video stream exclusively showing the guitar player, or a video stream that shows only the singer). Obviously it is important in these cases to keep the latency of the video signals under control, preferably as low as possible.
It is increasingly important that such solutions can be run on personal multipurpose devices (such as smartphones or tablets). Those devices are becoming a central point of communication for the users and they serve as a platform for the development of various extra functionalities, just by running software solutions on said devices. The same applies for wireless communication platforms. The importance of running solutions on widespread transport links (such as WiFi 802.11X) is growing, not just from a cost point of view, but also from a convenience point of view. Those devices are also easily serviceable, even from remote locations.
It is important to note that in those devices solutions exist that fulfil the above requirements, but to our knowledge, no solution running on those commodity platforms meets the needs of the described latency-constrained environment. In those environments the dynamic adaptation of the system to the performance available at any particular time suddenly becomes one of the most critical factors to take into account.
EP 2129170 is concerned with a scheme for creating a low latency audio transmission link. In a transmitter a stream of audio samples is coded and transmitted over a synchronous, low latency wireless link. In the receiver the inverse operations like decoding are performed to recover the audio stream. In this way a minimization of transmission latency is achieved while a relatively high audio quality is maintained. The proposed solution may be used for wireless audio transmission from an audio source to a listening device like a hearing instrument, e.g. from an audio source, such as a TV-set, to an intermediate device between the audio source and the listening device.
Application US2012/169588 deals with adjusting for input latency within an electronic device. The electronic device may receive a user input, such as a user actuation of a device key. A latency adjusted time of the input may be calculated based on a latency of the electronic device in determining the user actuation of the device key. The latency adjusted time may be used to determine a result of the user input.